mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-29 06:45:47 +00:00
aecd63478e
This is possible by not converting from LSBF to MSBF; instead add LSBF LUTs. This approach necessitates reversing the initial values. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
137 lines
4.4 KiB
C
137 lines
4.4 KiB
C
/*
|
|
* Direct Stream Digital (DSD) decoder
|
|
* based on BSD licensed dsd2pcm by Sebastian Gesemann
|
|
* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
|
|
* Copyright (c) 2014 Peter Ross
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
/**
|
|
* @file
|
|
* Direct Stream Digital (DSD) decoder
|
|
*/
|
|
|
|
#include "libavutil/mem.h"
|
|
|
|
#include "avcodec.h"
|
|
#include "codec_internal.h"
|
|
#include "decode.h"
|
|
#include "dsd.h"
|
|
|
|
#define DSD_SILENCE 0x69
|
|
#define DSD_SILENCE_REVERSED 0x96
|
|
/* 0x69 = 01101001
|
|
* This pattern "on repeat" makes a low energy 352.8 kHz tone
|
|
* and a high energy 1.0584 MHz tone which should be filtered
|
|
* out completely by any playback system --> silence
|
|
*/
|
|
|
|
static av_cold int decode_init(AVCodecContext *avctx)
|
|
{
|
|
DSDContext * s;
|
|
int i;
|
|
uint8_t silence;
|
|
|
|
if (!avctx->ch_layout.nb_channels)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
ff_init_dsd_data();
|
|
|
|
s = av_malloc_array(sizeof(DSDContext), avctx->ch_layout.nb_channels);
|
|
if (!s)
|
|
return AVERROR(ENOMEM);
|
|
|
|
silence = avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR ||
|
|
avctx->codec_id == AV_CODEC_ID_DSD_LSBF ? DSD_SILENCE_REVERSED : DSD_SILENCE;
|
|
for (i = 0; i < avctx->ch_layout.nb_channels; i++) {
|
|
s[i].pos = 0;
|
|
memset(s[i].buf, silence, sizeof(s[i].buf));
|
|
}
|
|
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
|
|
avctx->priv_data = s;
|
|
return 0;
|
|
}
|
|
|
|
typedef struct ThreadData {
|
|
AVFrame *frame;
|
|
const AVPacket *avpkt;
|
|
} ThreadData;
|
|
|
|
static int dsd_channel(AVCodecContext *avctx, void *tdata, int j, int threadnr)
|
|
{
|
|
int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR;
|
|
DSDContext *s = avctx->priv_data;
|
|
ThreadData *td = tdata;
|
|
AVFrame *frame = td->frame;
|
|
const AVPacket *avpkt = td->avpkt;
|
|
int src_next, src_stride;
|
|
float *dst = ((float **)frame->extended_data)[j];
|
|
|
|
if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) {
|
|
src_next = frame->nb_samples;
|
|
src_stride = 1;
|
|
} else {
|
|
src_next = 1;
|
|
src_stride = avctx->ch_layout.nb_channels;
|
|
}
|
|
|
|
ff_dsd2pcm_translate(&s[j], frame->nb_samples, lsbf,
|
|
avpkt->data + j * src_next, src_stride,
|
|
dst, 1);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int decode_frame(AVCodecContext *avctx, AVFrame *frame,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
ThreadData td;
|
|
int ret;
|
|
|
|
frame->nb_samples = avpkt->size / avctx->ch_layout.nb_channels;
|
|
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
return ret;
|
|
|
|
td.frame = frame;
|
|
td.avpkt = avpkt;
|
|
avctx->execute2(avctx, dsd_channel, &td, NULL, avctx->ch_layout.nb_channels);
|
|
|
|
*got_frame_ptr = 1;
|
|
return frame->nb_samples * avctx->ch_layout.nb_channels;
|
|
}
|
|
|
|
#define DSD_DECODER(id_, name_, long_name_) \
|
|
const FFCodec ff_ ## name_ ## _decoder = { \
|
|
.p.name = #name_, \
|
|
CODEC_LONG_NAME(long_name_), \
|
|
.p.type = AVMEDIA_TYPE_AUDIO, \
|
|
.p.id = AV_CODEC_ID_##id_, \
|
|
.init = decode_init, \
|
|
FF_CODEC_DECODE_CB(decode_frame), \
|
|
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SLICE_THREADS, \
|
|
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \
|
|
AV_SAMPLE_FMT_NONE }, \
|
|
};
|
|
|
|
DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")
|
|
DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first")
|
|
DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar")
|
|
DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar")
|